linux/sound/soc/pxa/tosa.c

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// SPDX-License-Identifier: GPL-2.0-or-later
/*
* tosa.c -- SoC audio for Tosa
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
* Richard Purdie <richard@openedhand.com>
*
* GPIO's
* 1 - Jack Insertion
* 5 - Hookswitch (headset answer/hang up switch)
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <mach/tosa.h>
#include <mach/audio.h>
#define TOSA_HP 0
#define TOSA_MIC_INT 1
#define TOSA_HEADSET 2
#define TOSA_HP_OFF 3
#define TOSA_SPK_ON 0
#define TOSA_SPK_OFF 1
static int tosa_jack_func;
static int tosa_spk_func;
static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
{
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
snd_soc_dapm_mutex_lock(dapm);
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_MIC_INT:
snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_HEADSET:
snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
}
static int tosa_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
/* check the jack status at stream startup */
tosa_ext_control(&rtd->card->dapm);
return 0;
}
static const struct snd_soc_ops tosa_ops = {
.startup = tosa_startup,
};
static int tosa_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = tosa_jack_func;
return 0;
}
static int tosa_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (tosa_jack_func == ucontrol->value.enumerated.item[0])
return 0;
tosa_jack_func = ucontrol->value.enumerated.item[0];
tosa_ext_control(&card->dapm);
return 1;
}
static int tosa_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = tosa_spk_func;
return 0;
}
static int tosa_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (tosa_spk_func == ucontrol->value.enumerated.item[0])
return 0;
tosa_spk_func = ucontrol->value.enumerated.item[0];
tosa_ext_control(&card->dapm);
return 1;
}
/* tosa dapm event handlers */
static int tosa_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0);
return 0;
}
/* tosa machine dapm widgets */
static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
SND_SOC_DAPM_HP("Headset Jack", NULL),
SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
};
/* tosa audio map */
static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to HPOUTL, HPOUTR */
{"Headphone Jack", NULL, "HPOUTL"},
{"Headphone Jack", NULL, "HPOUTR"},
/* ext speaker connected to LOUT2, ROUT2 */
{"Speaker", NULL, "LOUT2"},
{"Speaker", NULL, "ROUT2"},
/* internal mic is connected to mic1, mic2 differential - with bias */
{"MIC1", NULL, "Mic Bias"},
{"MIC2", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Mic (Internal)"},
/* headset is connected to HPOUTR, and LINEINR with bias */
{"Headset Jack", NULL, "HPOUTR"},
{"LINEINR", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Headset Jack"},
};
static const char * const jack_function[] = {"Headphone", "Mic", "Line",
"Headset", "Off"};
static const char * const spk_function[] = {"On", "Off"};
static const struct soc_enum tosa_enum[] = {
SOC_ENUM_SINGLE_EXT(5, jack_function),
SOC_ENUM_SINGLE_EXT(2, spk_function),
};
static const struct snd_kcontrol_new tosa_controls[] = {
SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
tosa_set_jack),
SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
tosa_set_spk),
};
SND_SOC_DAILINK_DEFS(ac97,
DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
SND_SOC_DAILINK_DEFS(ac97_aux,
DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
static struct snd_soc_dai_link tosa_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
.ops = &tosa_ops,
SND_SOC_DAILINK_REG(ac97),
},
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
.ops = &tosa_ops,
SND_SOC_DAILINK_REG(ac97_aux),
},
};
static struct snd_soc_card tosa = {
.name = "Tosa",
.owner = THIS_MODULE,
.dai_link = tosa_dai,
.num_links = ARRAY_SIZE(tosa_dai),
.controls = tosa_controls,
.num_controls = ARRAY_SIZE(tosa_controls),
.dapm_widgets = tosa_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.fully_routed = true,
};
static int tosa_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &tosa;
int ret;
ret = gpio_request_one(TOSA_GPIO_L_MUTE, GPIOF_OUT_INIT_LOW,
"Headphone Jack");
if (ret)
return ret;
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
gpio_free(TOSA_GPIO_L_MUTE);
}
return ret;
}
static int tosa_remove(struct platform_device *pdev)
{
gpio_free(TOSA_GPIO_L_MUTE);
return 0;
}
static struct platform_driver tosa_driver = {
.driver = {
.name = "tosa-audio",
.pm = &snd_soc_pm_ops,
},
.probe = tosa_probe,
.remove = tosa_remove,
};
module_platform_driver(tosa_driver);
/* Module information */
MODULE_AUTHOR("Richard Purdie");
MODULE_DESCRIPTION("ALSA SoC Tosa");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:tosa-audio");